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oza.4h07 at gmail.com
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 Posted: Thu Feb 18, 2016 6:29 am Post subject: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still |
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Hello,
I'm trying to have my first calls with WebRTC.
My server has asterisk 13.7.0.
I'm following the instructions from the wiki [1].
So I'm using [2] live demo from a Chrome navigator (v48) on Debian Jessie station.
Whenever I type something like ws://123.123.123.123:8088/ws in Expert Mode form (see [1]), I'm getting this error :
2:SecurityError: Failed to construct 'WebSocket': An insecure WebSocket connection may not be initiated from a page loaded over HTTPS.
If I replace ws://123.123.123.123:8088/ws with wss://123.123.123.123:8088/ws, this error message becomes with
Disconnected:Â Failed to connet to the server
My questions are:
1. Is wss now required by sipml5 live demo (implying wiki page is not up-to-date) ?
2. Do you have any pointer for WebRTC with Asterisk 13 and PJSIP ?
Regards
[1] https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5
[2] https://www.doubango.org/sipml5/ |
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simon.hohberg at mcs-d...
Guest
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| Posted: Thu Feb 18, 2016 7:31 am Post subject: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still |
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Hi Oliver,
On 02/18/2016 12:10 PM, Olivier wrote:
| Quote: | Hello,
I'm trying to have my first calls with WebRTC.
My server has asterisk 13.7.0.
I'm following the instructions from the wiki [1].
So I'm using [2] live demo from a Chrome navigator (v48) on Debian Jessie station.
Whenever I type something like ws://123.123.123.123:8088/ws in Expert Mode form (see [1]), I'm getting this error :
2:SecurityError: Failed to construct 'WebSocket': An insecure WebSocket connection may not be initiated from a page loaded over HTTPS.
If I replace ws://123.123.123.123:8088/ws with wss://123.123.123.123:8088/ws, this error message becomes with
Disconnected: Failed to connet to the server
My questions are:
1. Is wss now required by sipml5 live demo (implying wiki page is not up-to-date) ?
|
Yes, like the error says, you have to use wss on pages served via https. Furthermore, Chrome requires the use of https when you want to use getUserMedia.
See here: https://developers.google.com/web/updates/2015/10/chrome-47-webrtc?hl=en. It says: " Starting with Chrome 47, getUserMedia() requests are only allowed from secure origins: HTTPS or localhost."
The solution for development is, to host the webrtc client locally, so that you load the page from localhost. In that case getUserMedia is allowed with http, too (as the quote says). That means you have to download the dubango client and run a webserver on your dev machine.
| Quote: | 2. Do you have any pointer for WebRTC with Asterisk 13 and PJSIP ?
|
Unfortunately, there is not much documentation about this, as far as I can tell.
Regards,
Simon |
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oza.4h07 at gmail.com
Guest
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| Posted: Thu Feb 18, 2016 8:44 am Post subject: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still |
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Thank you much for yor reply.
2016-02-18 13:30 GMT+01:00 Simon Hohberg <simon.hohberg@mcs-datalabs.com (simon.hohberg@mcs-datalabs.com)>:
| Quote: | Hi Oliver,
On 02/18/2016 12:10 PM, Olivier wrote:
| Quote: | Hello,
I'm trying to have my first calls with WebRTC.
My server has asterisk 13.7.0.
I'm following the instructions from the wiki [1].
So I'm using [2] live demo from a Chrome navigator (v48) on Debian Jessie station.
Whenever I type something like ws://123.123.123.123:8088/ws in Expert Mode form (see [1]), I'm getting this error :
2:SecurityError: Failed to construct 'WebSocket': An insecure WebSocket connection may not be initiated from a page loaded over HTTPS.
If I replace ws://123.123.123.123:8088/ws with wss://123.123.123.123:8088/ws, this error message becomes with
Disconnected:Â Failed to connet to the server
My questions are:
1. Is wss now required by sipml5 live demo (implying wiki page is not up-to-date) ?
|
Yes, like the error says, you have to use wss on pages served via https. Furthermore, Chrome requires the use of https when you want to use getUserMedia.
See here: https://developers.google.com/web/updates/2015/10/chrome-47-webrtc?hl=en. It says: " Starting with Chrome 47, getUserMedia() requests are only allowed from secure origins: HTTPS or localhost."
|
Is it implied here that both HTTPS and WSS must also come from the same server (Same Origin Policy) ?
Then, can I also install my own WebRTC demo page on my own private Asterisk server and access this demo page through HTTPS ?
If I'm not mistaken, this should fulfill all requirements.
Â
| Quote: |
The solution for development is, to host the webrtc client locally, so that you load the page from localhost. In that case getUserMedia is allowed with http, too (as the quote says). That means you have to download the dubango client and run a webserver on your dev machine.
| Quote: | 2. Do you have any pointer for WebRTC with Asterisk 13 and PJSIP ?
|
Unfortunately, there is not much documentation about this, as far as I can tell.
|
I didn't find any.
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simon.hohberg at mcs-d...
Guest
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| Posted: Thu Feb 18, 2016 8:58 am Post subject: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still |
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| Quote: | Is it implied here that both HTTPS and WSS must also come from the
same server (Same Origin Policy) ?
|
No, the same origin policy does not apply to web sockets.
| Quote: | Then, can I also install my own WebRTC demo page on my own private
Asterisk server and access this demo page through HTTPS ?
If I'm not mistaken, this should fulfill all requirements.
|
It doesn't matter where the asterisk server is hosted. It is important
where the web application comes from. If you don't want to use https and
wss you only have the option to host the web app locally (on the same
machine as the browser that loads the page), which probably makes sense
only for development. Otherwise you have to use https and wss for the
reasons discussed earlier.
Hope it helps.
Simon
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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oza.4h07 at gmail.com
Guest
|
| Posted: Thu Feb 18, 2016 9:36 am Post subject: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still |
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2016-02-18 14:57 GMT+01:00 Simon Hohberg <simon.hohberg@mcs-datalabs.com (simon.hohberg@mcs-datalabs.com)>:
| Quote: |
| Quote: | Is it implied here that both HTTPS and WSS must also come from the same server (Same Origin Policy) ?
|
No, the same origin policy does not apply to web sockets.
| Quote: | Then, can I also install my own WebRTC demo page on my own private Asterisk server and access this demo page through HTTPS ?
If I'm not mistaken, this should fulfill all requirements.
|
It doesn't matter where the asterisk server is hosted. It is important where the web application comes from. If you don't want to use https and wss you only have the option to host the web app locally (on the same machine as the browser that loads the page), which probably makes sense only for development. Otherwise you have to use https and wss for the reasons discussed earlier.
Hope it helps. |
At least, it helped me to realize I still have several more things to learn 
My setup is the following:
- an asterisk server,
- a PC,
- asterisk server and PC are installed on the same LAN
- sipM5 live demo outside my LAN
- no NAT/PAT configuration allowing incoming communications from the outside.
Is using sipML live demo as a way to rapidly test private Asterisk WebRTC capabilies, something achievable ?
What would keep this from working ?
Â
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cervajs at fpf.slu.cz
Guest
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| Posted: Thu Feb 18, 2016 9:43 am Post subject: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still |
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my experience with pjsip for webrtc
http://lists.digium.com/pipermail/asterisk-users/2015-September/287562.html
Dne 18.2.2016 v 15:36 Olivier napsal(a):
| Quote: |
2016-02-18 14:57 GMT+01:00 Simon Hohberg <simon.hohberg@mcs-datalabs.com (simon.hohberg@mcs-datalabs.com)>:
| Quote: |
| Quote: | Is it implied here that both HTTPS and WSS must also come from the same server (Same Origin Policy) ?
|
No, the same origin policy does not apply to web sockets.
| Quote: | Then, can I also install my own WebRTC demo page on my own private Asterisk server and access this demo page through HTTPS ?
If I'm not mistaken, this should fulfill all requirements.
|
It doesn't matter where the asterisk server is hosted. It is important where the web application comes from. If you don't want to use https and wss you only have the option to host the web app locally (on the same machine as the browser that loads the page), which probably makes sense only for development. Otherwise you have to use https and wss for the reasons discussed earlier.
Hope it helps. |
At least, it helped me to realize I still have several more things to learn
My setup is the following:
- an asterisk server,
- a PC,
- asterisk server and PC are installed on the same LAN
- sipM5 live demo outside my LAN
- no NAT/PAT configuration allowing incoming communications from the outside.
Is using sipML live demo as a way to rapidly test private Asterisk WebRTC capabilies, something achievable ?
What would keep this from working ?
--
---------------------------------------
Marek Cervenka
=======================================
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oza.4h07 at gmail.com
Guest
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| Posted: Thu Feb 18, 2016 10:03 am Post subject: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still |
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2016-02-18 15:42 GMT+01:00 Marek ÄŒervenka <cervajs@fpf.slu.cz (cervajs@fpf.slu.cz)>:
Yes I saw this post earlier today.
Having to fight 14 days scared me a bit !
Did you set sipml5 on your own server or did you use Live demo (https://www.doubango.org/sipml5/call.htm?svn=241) ?
Â
| Quote: | Dne 18.2.2016 v 15:36 Olivier napsal(a):
| Quote: |
2016-02-18 14:57 GMT+01:00 Simon Hohberg <simon.hohberg@mcs-datalabs.com (simon.hohberg@mcs-datalabs.com)>:
| Quote: |
| Quote: | Is it implied here that both HTTPS and WSS must also come from the same server (Same Origin Policy) ?
|
No, the same origin policy does not apply to web sockets.
| Quote: | Then, can I also install my own WebRTC demo page on my own private Asterisk server and access this demo page through HTTPS ?
If I'm not mistaken, this should fulfill all requirements.
|
It doesn't matter where the asterisk server is hosted. It is important where the web application comes from. If you don't want to use https and wss you only have the option to host the web app locally (on the same machine as the browser that loads the page), which probably makes sense only for development. Otherwise you have to use https and wss for the reasons discussed earlier.
Hope it helps. |
At least, it helped me to realize I still have several more things to learn
My setup is the following:
- an asterisk server,
- a PC,
- asterisk server and PC are installed on the same LAN
- sipM5 live demo outside my LAN
- no NAT/PAT configuration allowing incoming communications from the outside.
Is using sipML live demo as a way to rapidly test private Asterisk WebRTC capabilies, something achievable ?
What would keep this from working ?
|
| Quote: | --
---------------------------------------
Marek Cervenka
=======================================
|
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
        http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
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cervajs at fpf.slu.cz
Guest
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| Posted: Fri Feb 19, 2016 6:01 am Post subject: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still |
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on my own server
i want try jssip
https://github.com/versatica/JsSIP
it looks like a lot "livelier" than sipml5
any experience with jssip?
Dne 18.2.2016 v 16:01 Olivier napsal(a):
| Quote: |
2016-02-18 15:42 GMT+01:00 Marek ÄŒervenka <cervajs@fpf.slu.cz (cervajs@fpf.slu.cz)>:
Yes I saw this post earlier today.
Having to fight 14 days scared me a bit !
Did you set sipml5 on your own server or did you use Live demo (https://www.doubango.org/sipml5/call.htm?svn=241) ?
Â
| Quote: | Dne 18.2.2016 v 15:36 Olivier napsal(a):
| Quote: |
2016-02-18 14:57 GMT+01:00 Simon Hohberg <[url=mailto:simon.hohberg@mcs-datalabs.com]simon.hohberg@mcs-datalabs.com (simon.hohberg@mcs-datalabs.com)[/url]>:
| Quote: |
| Quote: | Is it implied here that both HTTPS and WSS must also come from the same server (Same Origin Policy) ?
|
No, the same origin policy does not apply to web sockets.
| Quote: | Then, can I also install my own WebRTC demo page on my own private Asterisk server and access this demo page through HTTPS ?
If I'm not mistaken, this should fulfill all requirements.
|
It doesn't matter where the asterisk server is hosted. It is important where the web application comes from. If you don't want to use https and wss you only have the option to host the web app locally (on the same machine as the browser that loads the page), which probably makes sense only for development. Otherwise you have to use https and wss for the reasons discussed earlier.
Hope it helps. |
At least, it helped me to realize I still have several more things to learn
My setup is the following:
- an asterisk server,
- a PC,
- asterisk server and PC are installed on the same LAN
- sipM5 live demo outside my LAN
- no NAT/PAT configuration allowing incoming communications from the outside.
Is using sipML live demo as a way to rapidly test private Asterisk WebRTC capabilies, something achievable ?
What would keep this from working ?
|
|
--
---------------------------------------
Marek Cervenka
=======================================
|
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| Back to top |
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oza.4h07 at gmail.com
Guest
|
| Posted: Mon Feb 29, 2016 11:52 am Post subject: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still |
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2016-02-19 12:01 GMT+01:00 Marek ÄŒervenka <cervajs@fpf.slu.cz (cervajs@fpf.slu.cz)>:
Today, I'm back from holidays trip.
First of all, thanks for replying !
I'll try to use jssip as you suggested.
Anyway, I'm still failing to understand if wiki's page [1] is still valid with Asterisk 13, and if it's not valid anymore, which is the main change that prevent things to work.
[1] https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5
Â
| Quote: |
i want try jssip
https://github.com/versatica/JsSIP
it looks like a lot "livelier" than sipml5
any experience with jssip?
Dne 18.2.2016 v 16:01 Olivier napsal(a):
| Quote: |
2016-02-18 15:42 GMT+01:00 Marek ÄŒervenka <cervajs@fpf.slu.cz (cervajs@fpf.slu.cz)>:
Yes I saw this post earlier today.
Having to fight 14 days scared me a bit !
Did you set sipml5 on your own server or did you use Live demo (https://www.doubango.org/sipml5/call.htm?svn=241) ?
Â
| Quote: | Dne 18.2.2016 v 15:36 Olivier napsal(a):
| Quote: |
2016-02-18 14:57 GMT+01:00 Simon Hohberg < (simon.hohberg@mcs-datalabs.com)simon.hohberg@mcs-datalabs.com (simon.hohberg@mcs-datalabs.com)>:
| Quote: |
| Quote: | Is it implied here that both HTTPS and WSS must also come from the same server (Same Origin Policy) ?
|
No, the same origin policy does not apply to web sockets.
| Quote: | Then, can I also install my own WebRTC demo page on my own private Asterisk server and access this demo page through HTTPS ?
If I'm not mistaken, this should fulfill all requirements.
|
It doesn't matter where the asterisk server is hosted. It is important where the web application comes from. If you don't want to use https and wss you only have the option to host the web app locally (on the same machine as the browser that loads the page), which probably makes sense only for development. Otherwise you have to use https and wss for the reasons discussed earlier.
Hope it helps. |
At least, it helped me to realize I still have several more things to learn
My setup is the following:
- an asterisk server,
- a PC,
- asterisk server and PC are installed on the same LAN
- sipM5 live demo outside my LAN
- no NAT/PAT configuration allowing incoming communications from the outside.
Is using sipML live demo as a way to rapidly test private Asterisk WebRTC capabilies, something achievable ?
What would keep this from working ?
|
|
--
---------------------------------------
Marek Cervenka
=======================================
|
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
        http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
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jcolp at digium.com
Guest
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| Posted: Wed Mar 02, 2016 6:41 am Post subject: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still |
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Olivier wrote:
| Quote: |
2016-02-19 12:01 GMT+01:00 Marek ÄŒervenka <cervajs@fpf.slu.cz
<mailto:cervajs@fpf.slu.cz>>:
on my own server
Today, I'm back from holidays trip.
First of all, thanks for replying !
I'll try to use jssip as you suggested.
Anyway, I'm still failing to understand if wiki's page [1] is still
valid with Asterisk 13, and if it's not valid anymore, which is the main
change that prevent things to work.
|
If Chrome or the other browsers have changed things (or implemented new
requirements, ala HTTPS for serving stuff up) then it may not be correct
anymore. Chasing WebRTC is not currently something we currently do due
to the resources involved, but if the community can provide any changes
to the wiki page to help make it clearer or valid again they can be left
as a comment and we can incorporate them. If code changes are required
we do of course encourage those to be contributed[1].
Cheers,
[1] https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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oza.4h07 at gmail.com
Guest
|
| Posted: Wed Mar 02, 2016 12:45 pm Post subject: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still |
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I'm discovering WebRTC and I think it's a technology that is quite difficult to integrate with so many changing interfaces.
I think this is typically the kind of subject where the community could positively contribute to keep wiki pages updated.
As I'm quite interested in this topic, I'm assigning myself this task for the next weeks.
2016-03-02 12:40 GMT+01:00 Joshua Colp <jcolp@digium.com (jcolp@digium.com)>:
| Quote: | Olivier wrote:
| Quote: |
2016-02-19 12:01 GMT+01:00 Marek ÄŒervenka <cervajs@fpf.slu.cz (cervajs@fpf.slu.cz)
<mailto:cervajs@fpf.slu.cz (cervajs@fpf.slu.cz)>>:
  on my own server
Today, I'm back from holidays trip.
First of all, thanks for replying !
I'll try to use jssip as you suggested.
Anyway, I'm still failing to understand if wiki's page [1] is still
valid with Asterisk 13, and if it's not valid anymore, which is the main
change that prevent things to work.
|
If Chrome or the other browsers have changed things (or implemented new requirements, ala HTTPS for serving stuff up) then it may not be correct anymore. Chasing WebRTC is not currently something we currently do due to the resources involved, but if the community can provide any changes to the wiki page to help make it clearer or valid again they can be left as a comment and we can incorporate them. If code changes are required we do of course encourage those to be contributed[1].
Cheers,
[1] https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
       http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users
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